[asterisk-bugs] [JIRA] (ASTERISK-22644) Crash with app queue and DND set on SIP agent phone
Matt Jordan (JIRA)
noreply at issues.asterisk.org
Thu Oct 3 09:57:06 CDT 2013
[ https://issues.asterisk.org/jira/browse/ASTERISK-22644?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=210743#comment-210743 ]
Matt Jordan commented on ASTERISK-22644:
----------------------------------------
So I just tried reproducing this and can't.
My queue set up:
{noformat}
[sales]
strategy=ringall
announce=sales
musicclass = default
monitor-type=MixMonitor
monitor-format=wav
ringinuse = yes
weight = 0
wrapuptime = 0
{noformat}
Two dynamic agents login:
{noformat}
*CLI> == Using SIP RTP CoS mark 5
-- Executing [6000 at default:1] NoOp("SIP/digium02-00000007", "") in new stack
-- Executing [6000 at default:2] AddQueueMember("SIP/digium02-00000007", "sales") in new stack
[Oct 3 09:51:15] NOTICE[29070][C-00000001]: app_queue.c:7309 aqm_exec: Added interface 'SIP/digium02' to queue 'sales'
-- Executing [6000 at default:3] Hangup("SIP/digium02-00000007", "") in new stack
== Spawn extension (default, 6000, 3) exited non-zero on 'SIP/digium02-00000007'
== Using SIP RTP CoS mark 5
-- Executing [6000 at default:1] NoOp("SIP/digium01-00000008", "") in new stack
-- Executing [6000 at default:2] AddQueueMember("SIP/digium01-00000008", "sales") in new stack
[Oct 3 09:51:20] NOTICE[29071][C-00000002]: app_queue.c:7309 aqm_exec: Added interface 'SIP/digium01' to queue 'sales'
-- Executing [6000 at default:3] Hangup("SIP/digium01-00000008", "") in new stack
== Spawn extension (default, 6000, 3) exited non-zero on 'SIP/digium01-00000008'
{noformat}
This shows up in the sales queue:
{noformat}
*CLI> queue show sales
sales has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime, 0s talktime), W:0, C:0, A:1, SL:0.0% within 0s
Members:
SIP/digium02 (ringinuse enabled) (dynamic) (Not in use) has taken no calls yet
SIP/digium01 (ringinuse enabled) (dynamic) (Not in use) has taken no calls yet
No Callers
{noformat}
SIP/digium01 sets themselves to Busy. SIP/phone_a_2 then calls into the {{sales}} queue:
{noformat}
*CLI> == Using SIP RTP CoS mark 5
-- Executing [5000 at default:1] NoOp("SIP/phone_a_2-00000009", "") in new stack
-- Executing [5000 at default:2] Queue("SIP/phone_a_2-00000009", "sales,t,,") in new stack
-- Started music on hold, class 'default', on SIP/phone_a_2-00000009
== Using SIP RTP CoS mark 5
-- Called SIP/digium01
== Using SIP RTP CoS mark 5
-- Called SIP/digium02
-- SIP/digium02-0000000b connected line has changed. Saving it until answer for SIP/phone_a_2-00000009
-- SIP/digium01-0000000a connected line has changed. Saving it until answer for SIP/phone_a_2-00000009
-- Got SIP response 486 "Do Not Disturb" back from 10.24.19.80:5060
-- SIP/digium01-0000000a is busy
-- Nobody picked up in 0 ms
-- SIP/digium02-0000000b is ringing
-- Nobody picked up in 15000 ms
== Using SIP RTP CoS mark 5
-- Called SIP/digium01
== Using SIP RTP CoS mark 5
-- Called SIP/digium02
-- SIP/digium02-0000000d connected line has changed. Saving it until answer for SIP/phone_a_2-00000009
-- SIP/digium01-0000000c connected line has changed. Saving it until answer for SIP/phone_a_2-00000009
-- Got SIP response 486 "Do Not Disturb" back from 10.24.19.80:5060
-- SIP/digium01-0000000c is busy
-- Nobody picked up in 0 ms
-- SIP/digium02-0000000d is ringing
-- Nobody picked up in 15000 ms
== Using SIP RTP CoS mark 5
-- Called SIP/digium01
== Using SIP RTP CoS mark 5
-- Called SIP/digium02
-- SIP/digium02-0000000f connected line has changed. Saving it until answer for SIP/phone_a_2-00000009
-- SIP/digium01-0000000e connected line has changed. Saving it until answer for SIP/phone_a_2-00000009
-- Got SIP response 486 "Do Not Disturb" back from 10.24.19.80:5060
-- SIP/digium01-0000000e is busy
-- Nobody picked up in 0 ms
-- SIP/digium02-0000000f is ringing
-- Nobody picked up in 15000 ms
-- Stopped music on hold on SIP/phone_a_2-00000009
== Spawn extension (default, 5000, 2) exited non-zero on 'SIP/phone_a_2-00000009'
{noformat}
One thing that is different about your trace is that {{SIP/digium01}} is still dialed on the second time through. This probably keeps the interface from being NULL in the "last ditch effort" code at the end of Queue.
I noticed that your version of Asterisk is modified. Can you reproduce this with a fresh checkout of Asterisk that is unmodified?
> Crash with app queue and DND set on SIP agent phone
> ---------------------------------------------------
>
> Key: ASTERISK-22644
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-22644
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Applications/app_queue
> Affects Versions: 12.0.0-alpha1
> Environment: Centos 6.4 i386 + Grandstream GXP2000 phones
> Reporter: Marco Signorini
>
> I'm facing with a problem running a queue on Asterisk 12. It seems always replicable in my box.
> This is my setup:
> Asterisk SVN-branch-12-r400356M running on a CentOS 6.4 fresh install on bare Celeron machine.
> A queue defined as:
> {noformat}
> [300]
> ringinuse=yes
> strategy=ringall
> timeout=15
> weight=0
> wrapuptime=0
> {noformat}
> Two agents logged on that queue:
> {noformat}
> *CLI> queue show 300
> 300 has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s
> Members:
> SIP/204 (ringinuse enabled) (dynamic) (Not in use) has taken no calls yet
> SIP/200 (ringinuse enabled) (dynamic) (Not in use) has taken no calls yet
> No Callers
> {noformat}
> The SIP/204 is a Grandstream GXP2000 and has the DND set so it will never ring.
> As soon as a call enters in the queue, the SIP/200 starts ringing and the SIP/204 answers with a Busy back message. If the call is not answered, after the timeout period the PBX tries to contact the two agents but as soon as it contacts the SIP/204 it crashes.
> {noformat}
> *CLI> == Using SIP RTP CoS mark 5
> -- Executing [300 at from-internal:1] NoOp("SIP/201-00000000", ""Called Queue 300") in new stack
> -- Executing [300 at from-internal:2] Queue("SIP/201-00000000", "300,t,,") in new stack
> -- Started music on hold, class 'default', on SIP/201-00000000
> == Using SIP RTP CoS mark 5
> -- Called SIP/200
> == Using SIP RTP CoS mark 5
> -- Called SIP/204
> -- SIP/204-00000002 connected line has changed. Saving it until answer for SIP/201-00000000
> -- SIP/200-00000001 connected line has changed. Saving it until answer for SIP/201-00000000
> -- Got SIP response 486 "Busy" back from 10.10.5.117:5066
> -- SIP/204-00000002 is busy
> -- Nobody picked up in 0 ms
> -- SIP/200-00000001 is ringing
> -- SIP/200-00000001 is ringing
> -- Nobody picked up in 15000 ms
> -- Nobody picked up in 15000 ms
> Segmentation fault (core dumped)
> {noformat}
> I've generated a core dump and run the dgb to have the stack trace. Seems the PBX tries to manage a NULL channel.
> Here are the details:
> Core was generated by `asterisk -cvvvg'.
> Program terminated with signal 11, Segmentation fault.
> {noformat}
> #0 0x080f4a6f in ast_channel_tech (chan=0x0) at channel_internal_api.c:876
> 876 {
> Missing separate debuginfos, use: debuginfo-install glibc-2.12-1.107.el6_4.4.i686 libuuid-2.17.2-12.9.el6_4.3.i686 libxml2-2.7.6-12.el6_4.1.i686 ncurses-libs-5.7-3.20090208.el6.i686 nss-softokn-freebl-3.14.3-3.el6_4.i686 sqlite-3.6.20-1.el6.i686 zlib-1.2.3-29.el6.i686
> (gdb) bt
> #0 0x080f4a6f in ast_channel_tech (chan=0x0) at channel_internal_api.c:876
> #1 0x081d230d in ast_channel_snapshot_create (chan=0x0) at stasis_channels.c:184
> #2 0x00a6614f in queue_publish_multi_channel_blob (caller=<value optimized out>, agent=0x0, type=0x900491c, blob=0xb6b231a0) at app_queue.c:1955
> #3 0x00a77fc4 in rna (rnatime=15000, qe=0xb6867c88, peer=0x0, interface=0xb6b45964 "SIP/204", membername=0x8ff5c9c "SIP/204", autopause=1) at app_queue.c:4267
> #4 0x00a798b5 in wait_for_answer (qe=0xb6867c88, outgoing=0xb6b6f020, to=0xb6867bb8, digit=0xb6867bbf "", prebusies=0, caller_disconnect=0, forwardsallowed=1, ringing=0) at app_queue.c:4841
> #5 0x00a7b18a in try_calling (qe=0xb6867c88, opts=..., opt_args=0xb6868e04, announceoverride=0xb6867c27 "", url=0xb6867c26 "", tries=0xb6868e10, noption=0xb6868e0c, agi=0x0, macro=0x0, gosub=0x0, ringing=0) at app_queue.c:6239
> #6 0x00a7f25d in queue_exec (chan=0xb7421564, data=<value optimized out>) at app_queue.c:7624
> #7 0x08188a44 in pbx_exec (c=0xb7421564, app=0x8ff9c40, data=0xb6868ef8 "300,t,,") at pbx.c:1588
> #8 0x08193822 in pbx_extension_helper (c=0xb7421564, con=0xb6b7ef7a, context=0xb74221c8 "from-internal", exten=0xb7422218 "300", priority=2, label=0x0, callerid=0xb7407830 "201", action=E_SPAWN, found=0xb686b29c,
> combined_find_spawn=1) at pbx.c:4697
> #9 0x0819a1b1 in ast_spawn_extension (c=0xb7421564, args=0x0) at pbx.c:5706
> #10 __ast_pbx_run (c=0xb7421564, args=0x0) at pbx.c:6121
> #11 0x0819ba0d in pbx_thread (data=0xb7421564) at pbx.c:6440
> #12 0x081ede4a in dummy_start (data=0xb7422b58) at utils.c:1169
> #13 0x00285a49 in start_thread () from /lib/libpthread.so.0
> #14 0x00ec5aae in clone () from /lib/libc.so.6
> {noformat}
> The issue was already posted on the dev-list and this is the comment from Matthew Jordan:
> {quote}
> It's a crash, so it's clearly a bug. Please go ahead and file an issue in the issue tracker.
> Apparently o->chan can be NULL in the last ditch NOANSWER handler in wait_for_answer:
> {noformat}
> if (!*to) {
> for (o = start; o; o = o->call_next) {
> rna(orig, qe, o->chan, o->interface, o->member->membername, 1);
> }
> publish_dial_end_event(qe->chan, outgoing, NULL, "NOANSWER");
> }
> {noformat}
> Since rna does more than just raise the Stasis messages regarding the status of the queue, the code that raises the message should probably be made tolerant to a NULL peer channel.
> {quote}
> Thanks!
> Marco Signorini.
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